Prior systems and techniques for digitally encoding and decoding analog data or signals have been developed in the field of audio signal reproduction. Typically, such systems are based upon the Nyquist theorem which states that an analog bandwidth-limited signal can be reproduced exactly by sampling the signal at a rate at least twice the upper bandwidth limit of the signal to create a staircase approximation of the signal and by passing the staircase approximation through an ideal low-pass filter having a cut-off frequency equal to the upper bandwidth limit. In accordance with this theorem, prior systems have sampled the analog signal at a constant rate which is at least twice the highest anticipated frequency of the signal. For speech, the upper frequency limit is approximately 4 kilohertz while the upper frequency limit of music is considered to be the maximum frequency that can be detected by the human ear, i.e. 20 kilohertz. Accordingly, prior systems have used a sampling frequency of 44.1 kilohertz to encode musical signals.
The sampled analog signal is then converted into a series of digital signals which are stored on a storage medium, such as an audio compact disc.
During the reproduction process, the digital signals are sequentially retrieved from the storage medium at a constant rate, converted back to analog signals by a digital-to-analog converter and passed through a low-pass filter having a constant cut-off frequency equal to the highest anticipated frequency of the audio signal. In this manner, the analog signal is reconstructed without substantial distortion and loss.
While this type of encoding/decoding process results in highly accurate reproduction of the audio signal, it has been found that this process requires a large amount of digital signals to encode even a small portion of the audio signal. As a result, currently available compact discs are capable of recording only a limited amount (60-90 minutes) of stereo music.
Other systems have been devised which sample the analog signal at a variable rate wherein the rate is determined in dependence upon the magnitude of change of the amplitude of the analog signal. Such types of amplitude-dependent variable rate sampling, however, results in excessive oversampling of high-amplitude, low-frequency audio, music or voice signals. Such types of systems, therefore, also require a large amount of digital signals to encode such signals. Systems of this type are disclosed in Cherry et al U.S. Pat. No. 3,299,204, Stapleton U.S. Pat. No. 3,449,742, Jordan U.S. Pat. No. 4,308,585 and Kitamura U.S. Pat. No. 4,370,643.
A still different type of system for reproducing an analog signal is disclosed in Tsuchiya et al U.S. Pat. No. 4,348,699. This system includes means for establishing a sampling frequency at which an entire selected analog signal is sampled to create a digital signal and means for generating an encoded representation of the sampling frequency, which representation is recorded on a recording medium along with the digitally encoded analog signal. The representation of the sampling frequency is used to control the transport speed of the recording medium so that the system records the digital signals at a predetermined constant density on the recording medium. It does not appear, however, that this system results in a reduced or minimum amount of digital signals to encode an analog signal.
Kitamura et al U.S. Pat. No. 4,568,912 discloses a data compression system wherein a series of words which represent an analog signal at equally spaced intervals throughout the signal are converted into a second series of digital words which represent the analog signal at intervals between zero crossing points thereof. The intervals defined by the second series of words are equally spaced between successive zero crossing points of the analog signal but the interval duration in one portion of the signal between zero crossing points may be different than the duration of the intervals in another portion of the analog signal between two other zero crossing points.
The second series of words are analyzed to produce a straight line approximation of the analog signal and the approximation is filtered by a variable frequency low-pass filter to reconstruct the analog signal.
However, the Kitamura et al system does not attempt to reproduce high frequency components of an analog signal. Rather, it appears that the time between successive zero crossing points is detected and the rate at which the analog signal is sampled is determined in accordance with the detected time. Moreover, this system deliberately filters high frequency components from the original analog signal so that memory savings are achieved.
Also, it appears that the variable frequency low-pass filter described in the Kitamura et al patent is inherently slow in response time and hence could not reproduce high frequencies.